Audio signal component compensation system

ABSTRACT

A system compensating audio signal components in a communication system is disclosed, the method comprising the steps of detecting, by a microphone, a sound signal, the sound signal comprising audio signal components resulting from reproducing an audio signal of an audio source, the sound signal further comprising speech signal components corresponding to a speech signal from a person, filtering the sound signal to whiten the sound signal, compensating the audio signal components in the whitened sound signal, and removing the whitening of the compensated sound signal, where the filtering of the audio signal is performed using at least two filters in an alternating way, each filter using time-dependent filter coefficients.

RELATED APPLICATIONS

This application claims priority to European Patent Application SerialNo. 06 014 366.6, filed on Jul. 11, 2006, titled METHOD FOR COMPENSATIONOF AUDIO SIGNAL COMPONENTS IN A VEHICLE COMMUNICATION SYSTEM AND SYSTEMTHEREFOR, the application of which is incorporated by reference in itsentirety in this application.

BACKGROUND OF THE INVENTION

1. Field of the Invention

This invention relates to a communication system. More particularly, theinvention relates to a method and a system for compensation of audiosignal components in a communication system, such as in a vehiclecommunication system.

2. Related Art

The use of various types of communication systems has been proliferatingover the last few years. For example, communication systems are oftenincorporated in vehicles for different purposes. For example, it ispossible to use speech recognition and voice commands of the driver forcontrolling predetermined electronic devices inside the vehicle.Additionally, telephone calls, such as in a conference call, arepossible with two or more passengers within the vehicle. For example, aperson sitting on a front seat and a person sitting on one of the backseats may talk to a third person on the other end of the line using ahands-free communication system inside the vehicle. Moreover, it ispossible to use the communication system inside the vehicle for thecommunication of the different vehicle passengers to each other,

In vehicle communication systems, it may be difficult to hear speechaudibly and clearly due to noise, other sounds in the vehicle orattenuation of the speech sound waves. Accordingly, the voice of one ofthe passengers may be detected using one or more microphones positionedin different locations in the vehicle. The signal detected by themicrophone can be processed and then output using the loudspeakers of anaudio module that is normally located in the vehicle. The signal emittedfrom the loudspeaker, however, is normally also detected by themicrophone. To avoid acoustic feedback or other undesirable effects, thesignals detected by the microphone have to be processed and such signalcomponents have to be filtered out. Otherwise, undesirable feedback canoccur in the system.

In vehicle audio systems, it has become possible to select differentmodes for reproducing an audio signal. By way of example, state of theart audio systems provide the possibility to either reproduce the soundin a stereo mode or in a surround sound mode. In the surround soundmode, additional time delays may be introduced in the different audiochannels of the audio signal, so that the person sitting inside thevehicle has the impression of a surround sound audio system. When thisaudio system having a variable time delay in the different audiochannels is used in connection with a vehicle communication system, theaudio signal component emitted from the loudspeakers and then detectedby the microphone should be removed to avoid unwanted echoes. In asurround sound mode, the signal amplifier introduces an additional timedelay into the audio channel and the audio signal component detected bythe microphone is delayed by the time delay introduced by the amplifier.Accordingly, an echo compensation unit for compensating acoustic echoes,by simulating the signal path from the loudspeaker to the microphone,should be able to simulate this signal path with a variable time delay.For the echo compensation of audio signal components from a signaldetected by a microphone, a method with high computing power may benecessary. The required computer power mainly depends on the length ofthe filter of the echo compensation units. Thus generally, the greaterthe length of the filter, the more computer power needed.

Furthermore, it is possible that several microphones may be used for oneseat to detect the speech signal of a passenger. Negative feedback canbe avoided when adaptive filters are used for filtering out echoes andfeedback signal components of the signals.

In addition to the communication signals output via the loudspeakers ofthe vehicle, audio modules reproducing audio signals, such as radiosignals or signals from a music storage device such as a compact disc,are provided in the vehicles. These audio signals are output via thesame loudspeakers, and they are also recorded by the microphones andagain output via the loudspeaker. If these audio signal components arenot attenuated before being output as part of the signal detected by themicrophone, the driver has the impression of an audio sound signalhaving reverberation.

The above-described vehicle communication systems are often incorporatedinto expensive and highly sophisticated vehicles having highlysophisticated audio components. When the audio module is used inconnection with a vehicle communication system, the sound quality isdeteriorated by the feedback of the audio signal components picked up bythe microphone and again fed to the loudspeakers. To avoid this signalquality degradation, the audio signal may be disabled during thein-vehicle communication, or the audio signal components detected by themicrophone may be filtered out in an effective way.

For compensation of audio signal components in a sound signal (alsoreferred to as “echo compensation”), a filter may be used to simulatethe audio signal components of a sound signal that has been emitted fromthe loudspeaker and then detected by the microphone. However, the audiosignal component may be, for example, an audio signal of a classicalpiece of music, a pop piece of music, or perhaps an interview withoutmusic. For all these different kinds of music, the echo compensation mayhave to be carried out in a different way to be effective. The audiosignal components of the audio signal can have, in the case of a stereosignal for example, completely independent audio channels. In othersituations, such as, for example, in the case of speaking interviews orone speaking person, the two audio signal parts of the stereo signal maybe completely linear, depending on the signals. The echo compensationfor linearly dependent signals is a difficult task, as the adaptationalgorithms for calculating filter coefficients generally do not have awell-defined solution. When the audio signal changes from a piece ofmusic to a person speaking, it is desirable for the filters to beadapted to the new signal characteristics. This adaptation of the filtertales a certain amount of time and during this time unwanted echoes canoccur.

Moreover, echo compensation filters seek to simulate the path of thesound wave in the vehicle by calculating the pulse response. Theapproximation step may not result in a non-ambiguous and definiteanswer. Particularly in cases where the audio signal may be either amono signal or a multi-channel signal, the different channels beingcompletely linearly dependent from each other, a multi-channel stereoecho compensation filter may have the problem of finding the correctresult. In other words, the stereo echo compensation filter may not beable to accurately simulate the interior of the vehicle through whichthe sound passed before it is detected by the microphone in a correctway.

Accordingly, a need exists to effectively cope with the differentsituations that can occur in the compensation of audio signal componentsin an echo compensation unit, and generally for an improved system andmethod for compensation of audio signal components in a vehiclecommunication system. A need further exists to reduce the length offilters while maintaining a length sufficient to allow the echocompensation unit to be able to simulate the signal path of a stereosignal or of a signal in a surround sound mode. Yet a further needexists to effectively cope with the different situations that can occurin the compensation of audio signal components in an echo compensationunit.

SUMMARY

An echo compensation system for compensating audio signal components ina communication system is provided. The communication system may include(i) an audio unit for generating an audio signal, (ii) a microphone forreceiving a sound signal, (iii) a loudspeaker for outputting the soundsignal detected by the microphone and outputting the audio signalitself, (iv) an echo compensation unit for compensating the audio signalcomponents of the sound signal, and (v) a filter for whitening the soundsignal, the audio signal, or both signals. Applicants note that the term“sound signals” (which may also be referred to as “detected soundsignals”) refers to the signals detected by a microphone, including bothaudio signal components and speech signal components. The system mayfurther include at least two filters used in an alternating way forwhitening the sound signal, the audio signal, or both signals. Thesystem may further include a sound signal having different audiochannels, the time delay of the different audio channels relative toeach other being adjustable.

A calculating unit may also be provided for calculating time-dependentfilter coefficients. Additionally, a switch may be provided forswitching the supply of the time-dependent filter coefficients tovarious audio signal filters. Furthermore, a second switch may also beprovided to supply the simulated audio signal components to asubtracting unit, where the signal output from the echo compensationunit may be subtracted from the detected signal output. In addition, aninverse filter may be provided for removing the whitening of thewhitened error signal resulting in the echo compensated sound signal,where this inverse filter may also be connected to the calculating unit.

A method for compensating audio signal components in a communicationsystem is also provided. According to one implementation, a soundsignal, comprising audio signal components and speech signal components,is detected by a microphone. The detected sound signal is then filteredin order to whiten the sound signal. After whitening the detected soundsignal, the audio signal components in the sound signal are compensated.After compensation, the whitening of the compensated sound signal may beremoved.

According to another implementation, filter coefficients may becalculated and supplied to two audio filters in an alternating way, tobe used for whitening of signals. In such an implementation, thecalculated filter coefficients may be supplied to a first filter for afirst set of N cycles, and the calculated filter coefficients may besupplied to the other filter for a next set of N cycles resulting in arenewal of the filter coefficients of each filter every 2N cycles (i.e.,new filter coefficients for a given filter calculated every 2N cycles).

According to another implementation of the invention, a system andmethod for compensating audio signal components in a communicationsystem is provided using a mono echo compensation unit and amulti-channel (or stereo) echo compensation unit in combination. Theprovided echo compensation system may comprise a mono echo compensationunit for receiving one channel of an audio signal, and a multi-channelcompensation unit for receiving at least two channels of the audiosignal. When the audio signal changes its characteristic (for example,from music to a person speaking), either the mono echo compensation unitor the multi-channel echo compensation unit achieves the best echocompensation result. Accordingly, effective echo compensation can beachieved for any kind of audio signal.

According to yet another implementation, an echo compensation system isprovided that is able to suppress audio signal components of an audiosource having a variable time delay. In one implementation, theadaptation of the length of the variable time delay may be used alone,or in connection with other aspects or implementations of the invention.It is also possible that the variation of the length of the delayelement may be used in combination with the time-dependent filtercoefficients and/or in combination with the dual echo compensationstructure of a mono echo compensation unit in combination with amulti-channel echo compensation unit, as described above.

These and other objects, features and advantages of the presentinvention, as well as other devices, apparatuses, systems, methods,features and advantages of the invention, will be or will becomeapparent to one with skill in the art upon examination of the followingfigures and detailed description. It is intended that all suchadditional systems, methods, features and advantages be included withinthis description, be within the scope of the invention, and be protectedby the accompanying claims.

BRIEF DESCRIPTION OF THE FIGURES

The invention may be better understood by referring to the figuresdescribed below. The components in the figures are not necessarily toscale, emphasis instead being placed upon illustrating the principles ofthe invention. In the figures, like reference numerals designatecorresponding parts throughout the different views.

FIG. 1 shows one example of an implementation of an in-vehiclecommunication system in which an echo compensation system may be used.

FIG. 2 shows one example of an implementation of a system used forcompensating audio signal components in a communication system.

FIG. 3 shows a one example of an implementation of an echo compensationsystem in greater detail.

FIG. 4 is a flowchart illustrating a first example of an implementationof a method for compensating audio signal components in a communicationsystem.

FIG. 5 shows in further detail a flowchart comprising the steps forusing time-dependent filter coefficients for decorrelation.

FIG. 6 shows an example of an implementation of a dual echo compensationsystem in greater detail.

FIG. 7 is a flowchart illustrating another example of an implementationof a method for compensating audio signal components in a communicationsystem.

FIG. 8 shows pulse responses of an audio signal in a stereoamplification mode and in a surround sound mode.

FIG. 9 shows an echo compensation system introducing a variable timedelay during an echo compensation.

FIG. 10 shows the echo compensation system of FIG. 9 after changing thevariable time delay of the echo compensation.

DETAILED DESCRIPTION

While the present invention may be used in various types ofcommunication systems, the invention will be described below withspecific reference to an in-vehicle communication system as an exampleapplication of the invention.

FIG. 1 shows one example of an implementation of an in-vehiclecommunication system in which an echo compensation system may be used.Such an in-vehicle communication system may comprise a plurality ofloudspeakers 11 via which audio signals from an audio source unit 15 areomitted. In the vehicle, different passenger positions are possible. Forexample, the positions may include, without limitation, the position ofthe driver 12 a, the position of the front seat passenger 12 b, and twopositions in the back 12 c and 12 d. When one of the passengers in thefront 12 a, 12 b wants to communicate with one of the passengers sittingin the back 12 c, 12 d, or if two passengers, one in the front and onein the back, are communicating with a third person in atelecommunication system, one or more microphones 13 a-d may beprovided. For example, the microphones (or a ray or set of microphones)may include, without limitation, the following: a microphone 13 a fordetecting the speech signal of a passenger in the driver position 12 a,a microphone 13 b for detecting the speech signal of the front passenger12 b, a microphone 13 c for detecting the speech signal of the rearpassenger behind the driver 12 c, and a microphone 13 d for detectingthe speech signal of the rear passenger behind the front seat passenger12 d, may be provided. One of skill in the art would understand thatthese microphones 13 a-d may be positioned in other locations, or thatmore or fewer microphones may be used. For example, more than onemicrophone may be provided corresponding to a single passenger position.

When more than two microphones are used for one vehicle seat, a beamforming for the different vehicle seat positions can be done. In theexample implementation illustrated in FIG. 1, signals received from therear microphones 13 c-13 d may be supplied to a first signal processingunit 16 used for controlling the signal processing of speech signalsfrom the back seats 12 c-12 d to the front seats 12 a-12 b, and a signalprocessing unit 17 may receive signals from the front microphones 13a-13 b, and control the signal processing of speech signals from thefront seats 12 a-12 b to the back seats 12 c-12 d. In oneimplementation, the signal processing units 16 and 17 may determinethrough which loudspeakers 11 of the vehicle the signals detected by themicrophones 13 a-13 d will be output.

FIG. 2 shows one example of an implementation of a system used forcompensating audio signal components in a communication system. In FIG.2, an audio source unit 15 represents the audio signal source of FIG. 1having two different audio channels, a first channel x _(L)(n) and asecond channel x _(R)(n). While in this example a dual-channel audiosignal is shown, the system also applies to multiple channel audiosignals having more than two channels. The two audio signal channels(also referred to simply as audio signals) may then be transmitted to afilter unit 21 where they are either filtered in a time-variant manneror processed by a nonlinear characteristic to reduce the mutualcorrelation. This filtering is done to whiten or decorrelate the audiosignal components, as the echo compensation system may be more effectivewhen it is carried out on a whitened audio signal. A whitened signalgenerally indicates that the spectrum contains equal power per cycle,i.e., the signal has a flat spectrum that contains all differentfrequencies in equal amount. The filtering for whitening the audiosignal furthermore decorrelates the different channels of the audiosignal. One of skill in the art would understand that the filter unit 21is optional.

The filtered audio signal channels x_(L)(n) and x_(R)(n) are thentransmitted to an audio amplifier 22 for amplifying the audio signalsbefore they are emitted via the loudspeakers 11. The filtered audiosignal channels are also supplied to an echo compensation unit 23 wherethe audio signal components of a detected sound signal (not shown) maybe removed. The audio signals emitted from the loudspeakers 11 propagatein the environment and may be diffracted different times before they aredetected by one or more the microphones 13. The detected sound signal,comprising audio signal components as emitted by the loudspeaker 11 andalso comprising speech signal components (such as from one or more ofthe passengers) are then fed to a processing unit 24 where linearprocessing (beam forming etc.) of the detected sound signal can be done.The output signals of the two units 23 and 24 are then fed to asubtracting unit 25 where the signal output from the echo compensationunit 23, {circumflex over (d)}(n), is subtracted from the detectedsignal output from the processing unit 24, d(n). The subtraction resultsin an error signal as discussed further below. The better the echocompensation can simulate the signal path from the loudspeakers 11 tothe microphone 13, the smaller is the error signal e(n).

In the following, an example of the compensation of audio signalcomponents according to the implementation illustrated in FIG. 2 will bediscussed in more detail. The explanation is done on the basis of astereo signal source. However, the following explanation is also validfor an audio signal having multiple channels, such as five channels fora DVD. The radio signal of the left audio channel x_(L)(n) and of theright audio channel x_(R)(n) of the example stereo signal are output viaone or more loudspeakers 11 and reach the microphone(s) 13 after havingpassed the interior of the vehicle. The audio signal component detectedby the microphone(s) 13 comprises the direct audio signal as well assignal components diffracted, for example, by obstacles in the path ofthe sound signals. This signal transmission from the loudspeaker 11output to the microphone 13 as illustrated in FIG. 2 can be describedwith finite pulse responses:h _(L)(n)=[h _(L,0)(n),h _(L,1)(n), . . . , h _(L,L-1)(n)]^(T)   (1)h _(R)(n)=[h _(R,0)(n),h _(R,1)(n), . . . , h _(R,L-1)(n)]^(T)   (2)

The index n in equations (1) and (2) indicate the time dependence of thepulse responses. In one example, the signal path from the loudspeaker 11to the microphone 13 is simulated by filtering the audio signal in sucha way that after filtering, the filtered audio signal correspondssubstantially to the audio signal as it was detected by the microphone13. In this case, the unwanted audio signal component can be removedfrom the sound signal by subtracting the simulated audio signalcomponent from the detected sound signal.

For compensating the acoustic echoes, one or more adaptive filtershaving the following pulse responses can be used:ĥ _(L)(n)=[ĥ _(L,0)(n),ĥ _(L,1)(n), . . . , ĥ _(L,N-1)(n)]^(T)   (3)ĥ _(R)(n)=[ĥ _(R,0)(n),ĥ _(R,1)(n), . . . , ĥ _(R,N-1)(n)]^(T)   (4)

Normally, digital filters are used having a large number of filtercoefficients, e.g. 300-500 coefficients. The audio signal components asreceived by the microphones 13 can then be removed by subtracting thesimulated signal component from the detected sound signal. The resultingsignal is called an error signal e(n) and is defined as follows:$\begin{matrix}{{e(n)} = {{d(n)} - {\sum\limits_{i = 0}^{N - 1}{{{\hat{h}}_{L,i}(n)}{x_{L}\left( {n - i} \right)}}} - {\sum\limits_{i = 0}^{N - 1}{{{\hat{h}}_{R,i}(n)}{{x_{R}\left( {n - i} \right)}.}}}}} & (5)\end{matrix}$

The signal d(n) is either the signal from the microphone 13 or thesignal of a linear time invariant processing. A good compensation of theaudio signal component can be achieved when the estimated pulse responsecorresponds to the actual pulse responses and when a sufficient numberof coefficients are used. In echo compensation systems, the left and theright audio signal channels can have very different cross correlationcharacteristics. When music is reproduced as an audio sound signal, thesquare of the modulus of the coherence may be defined as:$\begin{matrix}{{C(\Omega)} = {\frac{S_{XLXR}(\Omega)}{\sqrt{{S_{XLXL}(\Omega)}{S_{XRXR}(\Omega)}}}}^{2}} & (6)\end{matrix}$

C(Ω) normally has values of C(Ω)<1. When reproducing a news signal orother signal comprising one speaker, the left and the right audiosignals may be linearly dependent signals, meaning that the coherence isapproximately 1. In the above-shown equation (6) the valuesS_(xLxR)(Ω),S_(xLxL)(Ω) and S_(xRxR)(Ω) are called the cross powerspectral density or auto power spectral density of the left and rightaudio signal channels x_(L)(n) and x_(R)(n). When one of the audiosignal components is an audio component that depends linearly on theother component, the adaptation algorithm compensating the acousticechoes may not have a non-ambiguous single solution.

FIG. 3 shows one implementation of an echo compensation system ingreater detail. In FIG. 3, the sound signal as detected by themicrophone 13 comprising the audio signal component and the speechsignal component is shown by y(n), and the audio signal itself (in thiscase, one channel of the audio signal) is represented by the signalx(n). In the example shown in FIG. 3, time-dependent decorrelationfilter coefficients are used. For calculating the time-dependentdecorrelation filter coefficients, a calculation unit 31 is providedwhere the time-dependent filter decorrelation coefficients arecalculated. The system of FIG. 3 may also include one or moredecorrelation filters 32, 33 a, 33 b for whitening the different signalcomponents. A first decorrelation filter 32 may be provided forwhitening the sound signal as detected by the microphone 13. Inaddition, decorrelation filters 33 a and 33 b may be provided forfiltering the audio signal itself. With decorrelated signals, it ispossible that the echo compensation can be carried out faster and in amore effective way.

In the example illustrated in FIG. 3, the audio signal x(n) may beprocessed in predetermined time intervals, and for each time intervalthe filter coefficients may be calculated. The filter coefficient of thefirst interval, e.g., an audio signal of 100 ms, once calculated by thecalculation unit 31, may be supplied to the first filter 33 a through aswitch 34. When the first filter 33 a has received a predeterminedamount of input samples (e.g., 500 samples), the switch 34 switches tothe second filter 33 b, and the calculated filter coefficientscalculated by calculation unit 31 are then transmitted to the otherdecorrelation filter 33 b. The switch 34 switches every N cycles, Nbeing the length of the echo compensation filters 35 a and 35 b. Duringthe time the filter coefficients are supplied to the first decorrelationfilter 33 a, the echo compensation filter 35 b may be used for theactual echo compensation. When the input samples for the echocompensation filter 35 a have been completely renewed, the switch 34changes its position and transmits the calculated filter coefficients tothe filter 33 b.

The audio signals are filtered by the echo compensation filters 35 a, 35b in such a way that the signal path in the vehicle is simulated. Theecho compensation filters 35 a, 35 b determine the pulse responsebetween the loudspeaker and the microphone. This can be done by usinggradient methods and using least mean square (LMS) algorithms ornormalized least mean square algorithms (NLMS). These methods andalgorithms are known in the art and will not be discussed in detail.

When the acoustic path of the vehicle is simulated in the echocompensation filters 35 a and 35 b, the output signal is then fed toanother switch 36, the switch 36 switching every N cycles, so that thefiltered signals from echo compensation filter 35 a are transmitted tothe subtracting unit 37 for N cycles, before the switch 36 is switchedand the signal from the echo compensation filter 35 b is fed to thesubtracting unit 37 for the next N cycles.

In the foregoing example, the two switches 34 and 36 change theirrespective states every N cycles, while at the same time eachrespectively maintaining a different actual state. Thus, when the switch34 supplies data to the upper branch 33 a and 35 a, the switch 36receives signal data from the lower branch 33 b and 35 b. In thisexample, the signal parameters in the filters 33 a and 33 b are renewedevery 2N cycles, where the signal parameters in the filter 32 arerenewed every N cycle. The output signal of filter 32 and the outputsignal of the echo compensation filters 35 a or 35 b are then used inthe subtracting unit where the simulated signal from the respective echocompensation filter 35 a, 35 b is subtracted from the filtered soundsignal as detected by the microphone 13. The result is a whitened errorsignal {tilde over (e)}(n). As it is known in adaptive filter systems,this whitened error signal {tilde over (e)}(n) is then used as afeedback control signal to adapt the audio signal echo compensationfilters. The whitened error signal {tilde over (e)}(n) is thentransmitted to an inverse filter 38 for removing the decorrelation. Thisinverse filter 38 also receives the calculated filter parameters every Ncycles. The resulting error signal e(n) output from the inverse filter38 then corresponds to the signal that will be output through theloudspeakers of the communication system. In this error signal e(n), theaudio signal component is removed or suppressed. With the system shownin FIG. 3, a changing audio signal source, such as a change from a pieceof music to a person speaking, can be detected within N cycles, and thedecorrelation filters can follow this change in music also in N cycles.

In the example shown in FIG. 3, the signal processing is shown for onechannel of the audio signal x(n). It should be understood that thisstructure of the two filter branches together with the two switches canbe applied for each audio channel or certain selected audio channels.Thus, the echo compensation system may comprise a plurality ofdecorrelation filters for whitening the audio signal and the soundsignal before the echo compensation, where one decorrelation filter isprovided for each channel of the audio signal. By way of example, asexplained above, the channel shown in FIG. 3 may be the left channel ofa stereo signal, and the right channel (x_(R)(n)) of the stereo audiosignal may utilize a second filter coefficient calculating unit havinganother two branches of filters. In this example, the filtered audiosignal for the right channel may then be combined with the filteredaudio signal for the left channel before the combined signal istransmitted to the subtracting unit 37. In the subtracting unit, thedetected sound signal comprises all of the individual audio channels,each channel having been processed as shown in FIG. 3, the differentchannels being combined before they are transmitted to the subtractingunit 37.

FIG. 4 is a flowchart illustrating an example of an implementation of amethod for compensating audio signal components in a communicationsystem. The method of FIG. 4 uses time-dependent filter coefficients.The method starts at step 41. First, an audio signal from an audiosource is output via the loudspeakers (step 42). When an in-vehiclecommunication system is used, a microphone may be provided for detectinga sound signal in the vehicle (step 43). The detected sound signal maycomprise components of the audio signal output via the loudspeakers, aswell as speech signal components corresponding to speech signals fromone or more passengers. Thus, the detected sound signal detected in step43 generally comprises two different components—the audio signalcomponent and a speech signal component. Next, the detected sound signaland the audio signal are whitened (step 44).

After whitening 44 (also referred to as decorrelating, since thewhitening of a signal decorrelates the different channels of thesignal), the acoustic echoes are compensated by compensating the audiosignal components in the sound signal (step 45). This compensation maybe carried out as explained in connection with FIG. 3 usingtime-dependent decorrelation filter coefficients and using alternatingcompensation units. Next, the whitening of the different signals isremoved in step 46 resulting in an improved error signal and the methodends at step 47.

FIG. 5 shows in further detail a flowchart comprising the steps forusing time-dependent filter coefficients for decorrelation. Inparticular, the alternating transmission of the filter coefficients forthe decorrelation filter is described in greater detail with respect toFIG. 5. As previously explained, according to this aspect of theinvention, the whitening of the audio signal may be performed using atleast two filters in all alternating way, each filter havingtime-dependent filter coefficients. When time-dependent filtercoefficients are used, the actual characteristic of the audio signal maybe taken into account. Accordingly, it is not necessary to use anaverage signal characteristic, as the filtering may be adapted to theactual audio signal. In this method, when one filter is being used forfiltering, the other filter may continue to receive the audio signal sothat filter coefficients for this new part of the audio signal can becalculated. With the use of time-dependent filter coefficients, theactual speed of the echo compensation filter compensating the audiosignal components can be improved. Furthermore, the use of two differentfilters in an alternating way may help to keep signal processing powerlow.

As illustrated in FIG. 5, the audio signal from the audio signal sourceis first supplied to a calculation unit 31 where the time-dependentfilter coefficients are calculated for the decorrelation filters every Ncycles (step 51). The filter coefficients are typically calculated basedon the audio signal itself (step 51), the filter coefficients beingrenewed every N cycles, N being the length of the compensation filter.By way of example, the length of the echo compensation filter may bechosen in such a way that it comprises 500 filter coefficients (i.e.N=500). Accordingly, in step 51, according to this example, thecalculated filter parameters are calculated by calculation unit 31 (seeFIG. 3) every 500 (“N”) cycles.

Next, the filter coefficients calculated by the calculation unit 31based, in this example, on the last 500 (N) cycles or input samples aretransmitted to the first decorrelation filter 33 a (step 52), which willuse and/or store this set of filter coefficients for 2N cycles. Duringthe time the filter coefficients are being calculated for thedecorrelation filter 33 a (i.e., the first N cycles), the other echocompensation filter 35 b is being used (step 52 a). The calculatedfilter coefficients calculated for the next N cycles are calculated instep 53 and are then transmitted to the other decorrelation filter 33 b(step 54). For this next N cycles during which new filter coefficientsare being calculated, the first echo compensation filter 35 a is used(step 54 a). In the method described with respect to FIG. 5, the audiosignal for a given decorrelation filter, once whitened or decorrelated,may then be supplied to a switch 37, the switch changing every N cyclesfrom one echo compensation filter to the other from where the signal istransmitted to the subtracting unit where it is subtracted from thewhitened sound signal.

When the filter coefficients are supplied to the first decorrelationfilter 33 a as shown in FIG. 3 (step 52), the filter coefficientscalculated the N cycles before are used for decorrelation and forcompensating the audio signal component in filter 33 b and 35 b. Theecho compensation filters 35 a and 35 b may each include a memorystorage unit in which the signals which were decorrelated with oldfilter parameters may be stored. When the filter parameters of thedecorrelation filters are changed, the decorrelation of the signal inthe echo compensation filters may be removed, and then the signal may bedecorrelated with the new filter parameters. For this kind of filtering,high computer power may be used to do the calculations. With the use oftwo different decorrelation filters and two different echo compensationfilters which are used in an alternating way the amount of computerpower required may be reduced.

FIG. 6 shows an example of an implementation of a dual echo compensationsystem in greater detail. The dual echo compensation system of FIG. 6uses two echo compensation units in combination—a mono echo compensationunit 62 and a multi-channel echo compensation unit 63. Generally, in theexample of FIG. 6, mono echo compensation and multi-channel or “stereo”echo compensation are carried out at the same time, and the compensationachieving the more desirable results is used. Again, the signal y(n) inthis example represents the sound signal detected by the microphones 13comprising the audio signal component and the speech signal component.The detected sound signal is supplied to a decorrelation filter 61 forwhitening the detected sound signal.

In the example of FIG. 6, echo compensation of a stereo signal is shown.The stereo signal has a first audio channel x_(L)(n) and the secondaudio channel x_(R)(n). These two signals are supplied to decorrelationfilters 61 for whitening the audio signal as was discussed in connectionwith FIG. 3. The whitened left audio signal is then input into a monoecho compensation unit 62 and to a stereo echo compensation unit 63. Themono echo compensation unit 62 comprises an echo compensation unit 621where the audio signal component of the sound signal as detected by themicrophone is simulated. The simulated audio signal is then input into asubtracting unit 622 where it is subtracted from the whitened soundsignal resulting in a whitened mono error signal {tilde over(e)}_(M)(n). The left audio channel is, after passing the decorrelationfilter 61, also input into the stereo echo compensation unit 63 where itis fed to an echo compensation unit 631 where the signal path issimulated as in the other echo compensation unit 621 and as described inconnection with FIGS. 1-5. Additionally, the whitened audio channel is,after passing the decorrelation filter 61, fed to a second signalcompensation unit 632. The output signals of the two echo compensationunits 631 and 632 are combined in the adder 635 before this combinedsignal is subtracted from the whitened sound signal in subtracting unit634. The output signal of the subtracting unit 634 is a whitened stereoerror signal {tilde over (e)}_(s)(n).

The system of FIG. 6 now has two output error signals, a mono errorsignal {tilde over (e)}_(M)(n) and a stereo error signal {tilde over(e)}_(s)(n). Depending on the actual composition of the audio signal,either the mono echo compensation unit or the stereo echo compensationunit achieves a more desirable result in removing the audio signalcomponent in the detected sound signal. When the audio signal is a monosignal or a linearly dependent stereo signal, the mono echo compensationunit will generally achieve the more desirable compensation results.Additionally, the mono echo compensation is generally faster. When theaudio signal is a stereo signal having non-linearly dependent signalcomponents, the stereo echo compensation unit will be able to compensateacoustic echoes. In order to compare the two signals, a comparison unit65 is provided having two inputs, one input being the output of the monoecho compensation unit {tilde over (e)}_(M)(n), one input being theoutput of the stereo echo compensation unit {tilde over (e)}_(s)(n).Comparison unit 65 compares the signal power of the two error signalsand selects the signal having the lower signal power as an output signal{tilde over (e)}(n). This output signal {tilde over (e)}(n) of thecomparison 65 unit is then transmitted to an inverse decorrelationfilter unit 66 removing the whitening of the echo compensated signal.The output error signal e(n) is then the signal that might be output bythe loudspeakers in which the audio signal components were effectivelyremoved.

The echo compensation unit shown in FIG. 6 can be single filterscompensating the echo. However, it is also possible to combine the monoand the multi-channel echo compensation with the time-dependent filtercoefficients described in connection with FIGS. 1-5. This means that foreach audio channel, a filter coefficient calculating unit such ascalculation unit 31 would be provided, and each of the echo compensationunits 621, 631 and 632 may be an echo compensation unit as shown in FIG.3 comprising a switch for supplying the calculated decorrelation filtercoefficients to one of the two branches of each echo compensation unit,another switch being provided for supplying the echo compensated signalto the subtracting unit. In this implementation of the invention, thetime-dependent filter coefficients would be combined with the mono andmulti-channel echo compensation units.

FIG. 7 is a flowchart illustrating another example of an implementationof a method for compensating audio signal components in a communicationsystem. According to the method of FIG. 7, a mono echo compensation unitand a multiple channel echo compensation unit are used in combination.The method of FIG. 7 starts at step 71. According to this method, theaudio signal is output via the loudspeaker in step 72. In step 73 asound signal is detected by the microphone, the sound signal having aspeech signal component and an audio signal component. In one example,the audio signal components may be removed in the detected sound signal,thus compensating any acoustic echoes. According to this example, thecompensation may comprise two different components. One channel of theaudio signal may be supplied to a mono echo compensation unit in step74, and in step 75 two or more channels of the multi-channel audiosignal are supplied to a multi-channel echo compensation unit. In bothecho compensation units, the echo compensation is carried out, be itwith time invariant decorrelation filter coefficients or be it inconnection with time-dependent decorrelation filter coefficients asdescribed in connection with FIGS. 1-5. In the next step 76, the outputof the mono echo compensation unit is compared to the output of themulti-channel echo compensation unit. In step 77, the signal outputhaving the lower signal power is selected and used as an echocompensated output signal of the sound signal detected by themicrophones 13. The method ends in step 78.

When a mono audio signal or a multi-channel (stereo) audio signal havingtwo linearly dependent signal channels is emitted through theloudspeakers, a mono echo compensation unit may achieve more desirableresults than a multi-channel stereo echo compensation unit. When thesound signal has non-linearly depending signal channels, the stereo echocompensation unit can compensate the audio signal components in thesound signal and therefore the acoustic echoes more effectively. As bothfilters in the example described with respect to FIGS. 6 and 7 are usedin parallel, the compensation unit having the more desirable result isselected. Thus, by using two different echo compensation units, anon-linear processing of the audio signals before the acoustic echoesare removed is not necessary, and the non-linear decorrelation of theaudio signals as a further step may be omitted. Moreover, this use oftwo different echo compensation units may improve signal quality.

Furthermore, in the case of a linearly dependent stereo signal or a monosignal, (e.g., an interview or other speech-only audio signal), the useof two different compensation units may increase the speed of echocompensation, as the mono echo compensation unit finds a solution in theapproximation method much faster than the multi-channel echocompensation unit. Further, when the audio signal changes, for example,from a piece of music to a person speaking, the echo compensation may beadapted more quickly with a mono and multi-channel echo compensationunit operating in parallel, than it would be if only a multi-channelecho compensation unit were used. Moreover, the output from the echocompensation unit that would achieve the best echo compensation result(e.g., the mono echo compensation unit or the multi-channel echocompensation unit) may be selected.

In accordance with the system described with respect to FIGS. 6 and 7,echo compensation may be carried out for each channel of an audio signalin the multi-channel echo compensation unit, the echo compensatedsignals of each channel being added before the resulting signal iscompared to the signal output of the mono echo compensation unit.Furthermore, before carrying out the echo compensation, a lineardecorrelation can be carried out for whitening the audio signal asdiscussed above. When the audio signal is a stereo signal, two channelsof the audio signal may be supplied to a multi-channel echo compensationunit, and one channel of the audio signal may be supplied to the monoecho compensation unit. Furthermore, the echo compensation may becarried out by simulating the audio signal components of the soundsignal as they are detected by the microphone 13 in the mono echocompensation unit and the multi echo compensation unit and bysubtracting the mono and the multi-channel simulated audio signalcomponents from the detected sound signal comprising both components.This subtraction results in a mono and a multi-channel error signal, thepower of the mono error signal and the power of the multi-channel errorsignal being compared in order to select the signal having the lowersignal power. In order to improve the echo compensation time-dependentfilter coefficients can be used for whitening the sound signal and forwhitening the audio signal as was discussed in connection with the firstaspect of the invention. Alternatively, the echoes may be compensated asdiscussed above in connection with the time-dependent filtercoefficients, with two different filters being used in an alternatingway as discussed above.

FIG. 8-10 illustrate a further aspect of one implementation of theinvention. FIG. 8 shows pulse responses of an audio signal in a stereoamplification mode and in a surround sound mode. Specifically, the uppergraph 81 of FIG. 8 illustrates a pulse response of a stereoamplification mode, and the lower part of FIG. 8 shows a graph 82 of apulse response of an audio signal in a surround sound mode. As can beseen by the comparison of the two graphs 81 and 82, an additional timedelay was introduced in the audio signal in the surround sound mode.

FIG. 9 shows part of an echo compensation system introducing a variabletime delay during an echo compensation. In the implementation of FIG. 9,a loudspeaker of the system may output the audio signal and the soundsignal received by the microphone or microphones. In general, aspreviously explained, an echo compensation unit may compensate acousticechoes by simulating the audio signal components in the sound signal asthey were detected by the microphone and by subtracting the simulatedaudio signal components from the detected sound signal. Also aspreviously explained, the echo compensation unit may comprise a filterfor filtering the audio signal to obtain the pulse response of the audiosignal. In addition to the filter, a delay element introducing avariable time delay into the audio signal before filtering may beprovided, a delay control unit being provided controlling the delayelement in such a way that the maximum of the pulse response is locatedwithin a predetermined range of filter coefficients of the filter. Thedelay element introducing a variable time delay into the audio signalbefore filtering allows keeping the length of the filter simulating theaudio signal component as received by the microphone 13, at a constantlength. The variable time delay introduced by the amplifier in thedifferent reproduction modes is introduced by the delay element. Thus,it is not necessary to provide a length of the filter that would be ableto simulate a maximum time delay introduced by the amplifier. This helpsto keep the computation time comparatively low.

According to one implementation of the invention, the delay elementcomprises a delay element 92 of variable length, the delay element ofvariable length being connected to a signal memory 93 of the filterfiltering the audio signal, the signal memory 93 of the filter having aconstant length. With the delay element 92 of variable length it ispossible to simulate the different time delays introduced by theamplifier of the audio signal. At the same time the signal memory 93 ofthe filter compensating the acoustic echoes can be of a relatively shortlength. In one example, the length of the delay element 92 is selectedin such a way that the maximum of the pulse response calculated by thefilter is located within a predetermined range of filter coefficients.

FIG. 9 will now be described in greater detail. In the upper part ofFIG. 9, graph 91 shows an example view of an audio signal. The echocompensation filter comprises a delay element 92 receiving an audiosignal or excitation signal 91. As previously stated, and as will bediscussed further below, the delay element 92 is of variable length. Thedelay element 92 introduces a variable delay before the audio signal istransmitted to a signal memory 93 of the echo compensation filter.Additionally, a memory 94 for storing the filter coefficients of theadaptive filter is provided. As it is known to those skilled in the art,different entries of the signal memory 93 are multiplied with the filtercoefficients and the different terms are added in an adder 96, resultingin an output signal of the adapted filter. Graph 95 shows the pulseresponse calculated by the filter. As can be seen by the indicated pulseresponse in graph 95, the maximum of the pulse response is located at afilter coefficient having a relatively large number.

At the beginning the filter coefficients are 0. This pulse response wascalculated based on the predetermined length of the delay memory. Above,the part 91 a of the audio signal 91 is shown, which is comprised in thedelay element 92. The other part 91 b of the audio signal 91 iscomprised in the signal memory 93 of the filter. With the length of thedelay element 92 shown in FIG. 9, a pulse response is calculated asshown by graph 95 having a maximum 95 a, which is located at a filtercoefficient having a relatively large number. When the pulse response 95is interpreted, one can deduce from the position of the maximum of thepulse response that the time delay introduced by the delay memory wasshorter than desired.

When it is detected that the maximum 95 a of the pulse response is notlocated at a predetermined filter coefficient, the pulse response isshifted as shown in FIG. 10. By shifting the pulse response as shown bygraph 105, so that the maximum 105 a is located at a predeterminedposition of the filter coefficients, the non-existing parts of the pulseresponse can be filled with zeroes as shown by the part 105 b of thegraph 105. In addition to the pulse response, the length of the delayelement 92 is also adjusted. In the example, shown the length of thedelay element 92 is increased, so that a larger part 91 c of the audiosignal is now comprised in the delay element 92, where only a smallerpart of the audio signal 91 d is now comprised in the signal memory 93of the filter. The new parts of the audio signal generated by theincreasing length of the delay element 92 can be filled with zeros asrepresented by part 91 e of the graph shown in FIG. 10. When comparingthe length of the respective delay elements 92 of FIGS. 9 and 10, it canbe deduced that by varying the length of the delay element 92, timedelays introduced in the different audio modes of an audio system can besimulated in an echo compensation unit. According to one implementationof the invention, the length of the delay element 92 can be controlledin such a way that the maximum of the pulse response is located at afilter coefficient which has a number around 30. It should be understoodthat any other number can be selected. However, the number of the filtercoefficient at which the maximum of the pulse response is to be locatedmay be selected in such a way that this filter coefficient is positionedat the beginning of the filter length. If the number is selected to betoo small, the system may not be able to precisely detect whether thedetermined maximum of the pulse response is actually the maximum orwhether the maximum is not represented in the filter coefficients. Byway of example, if it is detected that the maximum of the pulse responseis located within the first ten filter coefficients, it can be followedthat the time delay introduced by the delay element is larger thandesired. Accordingly, the length of the delay element 92 may beshortened and the impulse response may be shifted, i.e. the filtercoefficients in the coefficient memory 94 may be shifted. Again, theadded parts generated by the shifting are filled with zeroes.

This means that the direct sound as it is simulated by the echocompensation filter is situated at a predetermined filter coefficient ofthe filter. By way of example, the maximum of the pulse response can bearranged at a filter coefficient which is between one tenth and onetwentieth of the maximum filter coefficient. By way of example, it issupposed that the filter compensating the acoustic echoes has a lengthof 500 coefficients. In this example the delay element may be controlledin such a way that the maximum of the pulse response in the calculatedpulse response is positioned between the 20th and the 40th filtercoefficient, preferably between the 25th and 35th filter coefficient,even preferably between the 28th and the 32nd filter coefficient.

Preferably, the maximum of the pulse response can be calculated by thefollowing equation:i _(D)(n)=arg max(|h _(i)(n)|γ^(i)).   (7)

As can be seen by equation (7), the coefficient representing the directsound can be found by searching for the maximum of a weighted modulus ofthe pulse response. Preferably, the parameter γ is chosen to be between0 and 1. By introducing this parameter γ, reflections of the soundsignal may be attenuated relative to the direct sound. When the maximumof the pulse response in the simulated signal path in the echocompensation filter is found to be at a much larger filter coefficient,this means that the simulated time delay may be smaller than desired. Inthis case, a further time delay may be introduced. If, however, it isdetermined that the maximum of the pulse response is located at a filtercoefficient having a number which is smaller than the number of thepredetermined range, it can be followed that the simulated time delaymay be larger than desired. In this case, the delay introduced by thedelay element may be made shorter.

It should be understood that the implementations described in connectionwith FIGS. 9 and 10 can be combined with one of the implementationsdescribed in connection with FIGS. 1-5 and 6-7. It is also possible tocombine all three aspects of the invention, meaning that thetime-dependent decorrelation filter coefficients may be used incombination with the mono and multiple echo compensation units.Additionally, the echo compensation can be further improved by adjustingthe time delay as described in FIGS. 9 and 10. By way of example, whentime-dependent decorrelation filter coefficients are used, thecalculation of the time-dependent filter coefficients can be stoppedfrom time to time. When the calculation of the filter coefficients isstopped, the calculating power can be used to adapt the length of thedelay element by calculating the position of the maximum of the pulseresponse, by verifying whether this position is within a predeterminedrange and if not, by shifting the pulse response and by adapting thelength of the delay element accordingly.

Although the invention has been shown and described with respect toexample implementations thereof, it should be understood by thoseskilled in the art that the description is example rather than limitingin nature, and that many changes, additions and omissions are allpossible without departing from the scope and spirit of the presentinvention, which should be determined from the following claims.

1. A method for compensating an audio signal in a communication systemcomprising the steps of: detecting a sound signal, the sound signalcomprising a detected audio signal component from an audio signal and aspeech signal component; filtering the sound signal in order to create awhitened sound signal; filtering the audio signal component in order tocreate a whitened audio signal component, where the step of filteringthe audio signal component is performed using at least two filters in analternating way, each filter using time-dependent filter coefficients;compensating the audio signal component in the whitened sound signal;and inverse filtering the compensated whitened sound signal.
 2. Themethod of claim 1, further comprising the step of calculating thetime-dependent filter coefficients.
 3. The method of claim 2, where thestep of filtering the sound signal is performed using the time-dependentfilter coefficients.
 4. The method of claim 3, further comprising thestep of renewing the time-dependent filter coefficients every N cycles.5. The method of claim 3, further comprising alternately supplying thetime-dependent filter coefficients to a first filter and a secondfilter, where time-dependent filter coefficients calculated for a firstset of N cycles are supplied to the first filter, and time-dependentfilter coefficients calculated for a next set of N cycles are suppliedto the second filter, such that the time-dependent filter coefficientsfor the first filter and the second filter, respectively, are renewedevery 2N cycles.
 6. The method of claim 1, where the step ofcompensating the audio signal component in the whitened sound signalfurther comprises: simulating the whitened audio signal component tocreate a whitened simulated audio signal component; and determining awhitened error signal by subtracting the simulated audio signalcomponent from the sound signal.
 7. The method of claim 6, furthercomprising determining an estimated sound signal component using theerror signal as a feedback control signal.
 8. The method of claim 7,where the step of simulating the audio signal component of the soundsignal is performed by an echo compensation filter having a length equalto N.
 9. The method of claim 6, where the step of determining a whitenederror signal comprises alternately subtracting a first whitenedsimulated audio signal component from the whitened sound signal, and asecond whitened simulated audio signal component from the whitened soundsignal.
 10. The method of claim 9, further comprising inverse filteringthe whitened simulated error signal, resulting in an error signalcorresponding to an echo compensated sound signal.
 11. The method ofclaim 9, where the step of determining a whitened error signal comprisesalternately subtracting a first whitened simulated audio signalcomponent from the whitened sound signal for a first set of N cycles,and a second whitened simulated audio signal component from the whitenedsound signal for a next set of N cycles.
 12. A method for compensatingan audio signal comprising the steps of: detecting a sound signal, thesound signal comprising a detected audio signal component from an audiosignal, and a speech signal component; calculating time dependent filtercoefficients; filtering the sound signal to create a whitened soundsignal using a decorrelation filter, said decorrelation filter beingsupplied with the time dependent filter coefficients every N cycles;filtering the audio signal component to create a whitened audio signalcomponent where the step of filtering the audio signal component isperformed using at least two filters in an alternate way, said twofilters being supplied alternately every 2N cycles with said timedependent filter coefficients; compensating the audio signal componentin the whitened sound signal; and inverse filtering the compensatedwhitened sound signal.
 13. An echo compensation system comprising: atleast one microphone for detecting a sound signal, the sound signalcomprising a detected audio signal component from an audio signal and aspeech signal component; at least one loudspeaker for outputting thesound signal and the audio signal; a filter unit comprising a firstfilter and a second filter used in an alternating way for filtering theaudio signal component, where each filter uses time-dependent filtercoefficients; and an echo compensation unit for compensating the audiosignal component received by the microphone.
 14. The echo compensationsystem of claim 13, further comprising a calculating unit forcalculating the time-dependent filter coefficients.
 15. The echocompensation system of claim 14, further comprising a first switch incommunication with the calculating unit and the first and secondfilters, for alternately supplying the time-dependent filtercoefficients to the first filter and the second filter.
 16. The echocompensation system of claim 15, where the first switch switches betweena first position and a second position every N cycles.
 17. The echocompensation system of claim 13, where the filter unit further comprisesa third filter for filtering the sound signal and outputting a whitenedsound signal component, where the third filter receives thetime-dependent filter coefficients calculated by the calculating unit,and the time-dependent filter coefficients received by the third filterare refreshed every N cycles.
 18. The echo compensation system of claim13, where the echo compensation unit comprises a first echo compensationfilter in communication with the first filter and a second echocompensation filter in communication with the second filter, where thefirst echo compensation filter receives a first whitened audio signalcomponent from the first filter and outputs a first whitened simulatedaudio signal component, and the second echo compensation filter receivesa second whitened audio signal component from the second filter andoutputs a second whitened simulated audio signal component.
 19. The echocompensation system of claim 18, where the echo compensation unitfurther comprises a subtracting unit where the first and second whitenedsimulated audio signal components are subtracted from the whitened soundsignal component, resulting in a whitened error signal.
 20. The echocompensation unit of claim 19, where the whitened error signal is usedas a feedback control signal for the echo compensation filters.
 21. Theecho compensation unit of claim 13, further comprising an inverse filterfor inverse filtering the whitened error signal, and outputting an echocompensated sound signal.
 22. The echo compensation unit of 13, furthercomprising a third filter and a fourth filter, where the first and thirdfilters correspond to a first audio signal component, and the second andfourth filters correspond to a second audio signal component; and fourecho compensation filters, where two of the echo compensations filterscorrespond to the first audio signal component and the other two echocompensation filters correspond to the second audio signal component.23. The echo compensation unit of claim 18, further comprising a switchfor alternately supplying the first whitened simulated audio signalcomponent and the second whitened simulated audio signal component tothe subtracting unit, the switch alternating every N cycles.
 24. An echocompensation system comprising: at least one microphone for detecting asound signal, the sound signal comprising a detected audio signalcomponent from an audio signal and a speech signal component; at leastone loudspeaker for outputting the sound signal and the audio signal; acalculating unit for calculating time dependent filter coefficientsbased on the audio signal; at least two filters for alternatelyfiltering the audio signal, said filters being alternately supplied withsaid time-dependent filter coefficients every 2N cycles; a sound signalfilter for filtering the sound signal, the sound signal filter beingsupplied with said time dependent filter coefficients every N cycles; anecho compensation unit for compensating the audio signal component. 25.A method for compensating audio signal components comprising the stepsof: detecting a sound signal, the sound signal comprising a detectedaudio signal component from an audio signal comprising a first channeland a second channel, and a speech signal component; generating an echocompensated sound signal to compensate acoustic echoes in the soundsignal due to the detected audio signal component in the sound signal,where the generating step comprises the steps of: supplying the firstchannel of the audio signal to a mono echo compensation unit; supplyingthe first and second channels of the audio signal to a multi channelecho compensation unit; outputting a first output associated with afirst signal power from the mono echo compensation unit, and a secondoutput associated with a second signal power from the multi channel echocompensation unit; comparing the first signal power and the secondsignal power; selecting the first output if the first signal power issmaller than the second signal power; and selecting the second output ifthe second signal power is smaller than the first signal power.
 26. Themethod of claim 25, further comprising the step of filtering the soundsignal in order to obtain a whitened sound signal before the step ofgenerating the echo compensated sound signal, and inverse filtering theselected output.
 27. The method of claim 25, where the step ofgenerating an echo compensated sound signal further comprises:generating a first simulated audio signal component for the firstchannel and a second simulated audio signal component for the secondchannel using the multi channel echo compensation unit; and adding thefirst and second simulated audio signals to obtain a combined simulatedaudio signal component.
 28. The method of claim 26, further comprisingcalculating time-dependent filter coefficients to be used for obtainingthe whitened sound signal.
 29. The method of claim 27, furthercomprising subtracting a mono simulated audio signal component from thesound signal to obtain the first output, and subtracting the combinedsimulated audio signal component from the sound signal to obtain thesecond output.
 30. A method for compensating audio signal componentscomprising the steps of: detecting a sound signal, the sound signalcomprising a detected audio signal component from an audio signalcomprising a first channel and a second channel, and a speech signalcomponent; filtering the sound signal to obtain a whitened sound signal;filtering the first channel to obtain a first whitened audio signalcomponent; filtering the second channel to obtain a second whitenedaudio signal component; supplying the first whitened audio signalcomponent and the whitened sound signal to a mono echo compensationunit; outputting a first output having a first signal power from themono echo compensation unit; supplying the first whitened audio signalcomponent, the second whitened audio signal component, and the whitenedsound signal to a multi channel echo compensation unit; outputting asecond output having a second signal power from the multi channel echocompensation unit; comparing the first signal power and the secondsignal power; selecting the first output if the first signal power issmaller than the second signal power; and selecting the second output ifthe second signal power is smaller than the first signal power.
 31. Anecho compensation system comprising: at least one microphone fordetecting a sound signal, the sound signal comprising a detected audiosignal component from an audio signal comprising a first channel and asecond channel, and a speech signal component; at least one loudspeakerfor outputting the sound signal; a mono echo compensation unit forreceiving the first channel of the audio signal and outputting firstoutput having a first signal power; a multi channel echo compensationunit for receiving the first and second channels of the audio signal andoutputting a second output having a second signal power; and acomparison unit for comparing the first signal power and the secondsignal power; and selecting the first output if the first signal poweris lower than the second signal power, or the second output if thesecond signal power is lower than the first signal power.
 32. The echocompensation system of claim 31, further comprising a plurality offilters to whiten the audio signal and the sound signal, and an inversefilter for inverse filtering at least one of the first output and thesecond output.
 33. The echo compensation system of claim 32, where theplurality of filters includes at least one filter for the first channeland at least one filter for the second channel.
 34. An echo compensationsystem comprising: at least one microphone for detecting a sound signal,the sound signal comprising a detected audio signal component from anaudio signal comprising a first channel and a second channel, and aspeech signal component; at least one loudspeaker for outputting thesound signal; a filter unit for generating a whitened sound signal; aplurality of filter units for generating a whitened audio signal, thewhitened audio signal comprising a first whitened audio signalcorresponding to the first channel and a second whitened audio signalcorresponding to the second channel; a mono echo compensation unit beingsupplied with the first whitened audio signal and with the whitenedsound signal, and outputting a first output; a multi channel echocompensation unit being supplied with the first whitened audio signal,the second audio signal, and the whitened sound signal, and outputting asecond output; and a comparison unit for comparing a signal power of thefirst output and a signal power of the second output, and selectingwhichever of the first output or second output has a lower signal power.35. An echo compensation system comprising: an audio source forgenerating an audio signal having a first channel with a first timedelay and a second channel with a second time delay, where the first andsecond time delays are adjustable relative to each other; at least onemicrophone for detecting a sound signal, the sound signal comprising adetected audio signal component from the audio signal, and a speechsignal; a loudspeaker unit for outputting the audio signal and the soundsignal; an echo compensation unit for simulating the audio signalcomponent to obtain a simulated audio signal component, and subtractingthe simulated audio signal component from the sound signal, the echocompensation unit comprising: a filter for filtering the audio signal toobtain a pulse response of the audio signal, the pulse response having amaximum value; a plurality of filter coefficients corresponding to thefilter; a delay element for introducing a variable time delaycorresponding to the audio signal; and a delay control unit forcontrolling the delay element so that the maximum value of the pulseresponse is located within a predetermined range of the filtercoefficients.
 36. The echo compensation system of claim 35, where thedelay element comprises a delay memory having a variable length.
 37. Theecho compensation system of claim 36, where the delay element is incommunication with a signal memory, the signal memory of the filterhaving a constant length.
 38. The echo compensation system of claim 35,where a filter coefficient corresponding to the maximum value of thepulse response is located between a tenth and a twentieth filtercoefficient.
 39. The echo compensation system of claim 35, where themaximum value of the pulse response is located between a twentieth and afortieth filter coefficient
 40. The echo compensation system of claim35, where the maximum value of the pulse response is located between atwenty-fifth and a thirty-fifth filter coefficient.
 41. The echocompensation system of claim 35, where the maximum value of the pulseresponse is located between a twenty-eighth and a thirty-second filtercoefficient.
 42. The echo compensation system of claim 35, where thedelay control unit determines at which filter coefficient the maximumvalue of the impulse response is positioned.
 43. The echo compensationsystem of claim 35, further comprising: a filter for generating awhitened sound signal using time-dependent filter coefficients; aplurality of filter units for generating a whitened delayed audiosignal, where at least one filter unit comprises two audio signalfilters, each audio signal filter using time-dependent filtercoefficients for generating corresponding whitened delayed audiosignals, said at least two audio signal filters being used in analternating way for generating the whitened delayed audio signals; amono echo compensation unit comprising at least two echo compensationfilters alternately receiving the whitened delayed audio signals fromthe audio signal filters, and outputting a mono compensated sound signalhaving a first signal power; a multi channel echo compensation unitcomprising at least two echo compensation filters alternately receivingthe whitened delayed audio signals from the audio signal filters, andoutputting a multi channel compensated sound signal having a secondsignal power; and a comparison unit for comparing the first signal powerand the second signal power, and selecting whichever of the monocompensated sound signal and the multi channel compensated sound signalhas a lower signal power.
 44. An echo compensation system comprising: anaudio source for generating an audio signal having a first channel witha first time delay and a second channel with a second time delay, wherethe first and second time delays are adjustable relative to each other;at least one microphone for detecting a sound signal, the sound signalcomprising a detected audio signal component from the audio signal, anda speech signal; a loudspeaker unit for outputting the audio signal andthe sound signal; a delay element comprising a delay memory forgenerating a delayed audio signal; a calculating unit for calculating apulse response corresponding to the delayed audio signal and the soundsignal, the pulse response having a maximum value; an echo compensationunit for obtaining a compensated audio signal component and a delaycontrol unit for controlling the delay element so that the maximum valueof the pulse response is located within a predetermined range of filtercoefficients.
 45. A method for compensating audio signal componentscomprising the steps of: reproducing an audio signal, the audio signalhaving a first channel with a first time delay and a second channel witha second time delay, where the first and second time delays areadjustable relative to each other; outputting the audio signal;detecting a sound signal, the sound signal comprising a detected audiosignal component from the audio signal, and a speech signal component;outputting the sound signal; generating a simulated audio signalcomponent; subtracting the simulated audio signal components from thesound signal; simulating a signal path of the audio signal from theloudspeaker to the microphone by determining a pulse response of theaudio signal; the pulse response having a maximum value; and introducinga variable time delay selected so that the maximum value of the pulseresponse is located within a predetermined range of filter coefficientsof the filter.
 46. The method of claim 45, further comprising the stepof adding a variable time delay to the audio signal by supplying theaudio signal to a delay element of variable length.
 47. The method ofclaim 45, further comprising the step of determining the maximum valueof the pulse response and determining at which filter coefficient thedetermined maximum value is located.
 48. The method of claim 47 wherethe step of determining the maximum value of the pulse responsecomprises determining the maximum value of a weighted modulus of thepulse response.
 49. The method of claim 45, further comprising varyingthe length of a delay element.
 50. The method of claim 49, furthercomprising increasing the delay element when it is determined that themaximum value of the pulse response is positioned at a filtercoefficient having a number larger than the predetermined range.
 51. Themethod of claim 50, further comprising decreasing the delay element whenit is determined that the maximum value of the pulse response ispositioned at a filter coefficient having a number smaller than thepredetermined range.
 52. The method of claim 45, further comprisingshifting the pulse response so that the maximum of the pulse response islocated within the predetermined range of filter coefficients, if thedetermined pulse response is not located within a predetermined range offilter coefficients.
 53. The method of claim 45, further comprising thesteps of: filtering the sound signal to obtain a whitened sound signal;filtering the first channel and the second channel to obtain a firstwhitened delayed channel and a second whitened delayed channel, wherethe filtering of the first and second channels is performed using atleast two audio filters per channel in an alternating way, each audiofilter using time-dependent filter coefficients; obtaining a compensatedmono sound signal having a first signal power using a mono echocompensation unit; obtaining a compensated multi channel sound signalhaving a second signal power using a multi channel echo compensationunit; comparing the compensated first signal power and the second signalpower; and selecting whichever of the mono sound signal and thecompensated multi channel sound signal has a lower signal power.
 54. Amethod for compensating audio signal components comprising the steps of:reproducing an audio signal, the audio signal having a first channelwith a first time delay and a second channel with a second time delay,where the first and second time delays are adjustable relative to eachother; outputting the audio signal; detecting a sound signal, the soundsignal comprising a detected audio signal component from the audiosignal, and a speech signal component; outputting the sound signal;delaying the audio signal using a delay element; calculating a pulseresponse from the delayed audio signal and the sound signal, the pulseresponse having a maximum value; generating a simulated audio componentusing the delayed audio signal and a filter, said filter using thecalculated pulse response as filter coefficients; controlling a delayintroduced by the delay element so that the maximum value of the pulseresponse is positioned within a predetermined range of filtercoefficients; and subtracting the simulated audio component from thesound signal.